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https://github.com/xenia-project/FFmpeg.git
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ffplay: estimate audio frame pts from the previous frame pts values
Previously we estimated the audio packet pts instead of the frame pts, therefore it only worked within a single packet (containing multiple frames). The new method works accross seperate audio packets as well and also handles better the case if a decoder buffers several packets before outputting a decoded frame. Signed-off-by: Marton Balint <cus@passwd.hu>
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parent
782e06e292
commit
b383498ea8
11
ffplay.c
11
ffplay.c
@ -216,6 +216,7 @@ typedef struct VideoState {
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int frame_drops_early;
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int frame_drops_late;
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AVFrame *frame;
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int64_t audio_frame_next_pts;
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enum ShowMode {
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SHOW_MODE_NONE = -1, SHOW_MODE_VIDEO = 0, SHOW_MODE_WAVES, SHOW_MODE_RDFT, SHOW_MODE_NB
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@ -2158,6 +2159,8 @@ static int audio_decode_frame(VideoState *is)
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break;
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}
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pkt_temp->dts =
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pkt_temp->pts = AV_NOPTS_VALUE;
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pkt_temp->data += len1;
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pkt_temp->size -= len1;
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if (pkt_temp->data && pkt_temp->size <= 0 || !pkt_temp->data && !got_frame)
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@ -2171,8 +2174,11 @@ static int audio_decode_frame(VideoState *is)
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is->frame->pts = av_rescale_q(is->frame->pts, dec->time_base, tb);
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else if (is->frame->pkt_pts != AV_NOPTS_VALUE)
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is->frame->pts = av_rescale_q(is->frame->pkt_pts, is->audio_st->time_base, tb);
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if (pkt_temp->pts != AV_NOPTS_VALUE)
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pkt_temp->pts += (double) is->frame->nb_samples / is->frame->sample_rate / av_q2d(is->audio_st->time_base);
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else if (is->audio_frame_next_pts != AV_NOPTS_VALUE)
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is->frame->pts = av_rescale_q(is->audio_frame_next_pts, (AVRational){1, is->audio_filter_src.freq}, tb);
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if (is->frame->pts != AV_NOPTS_VALUE)
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is->audio_frame_next_pts = is->frame->pts + is->frame->nb_samples;
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#if CONFIG_AVFILTER
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dec_channel_layout = get_valid_channel_layout(is->frame->channel_layout, av_frame_get_channels(is->frame));
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@ -2326,6 +2332,7 @@ static int audio_decode_frame(VideoState *is)
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if (pkt->data == flush_pkt.data) {
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avcodec_flush_buffers(dec);
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is->audio_buf_frames_pending = 0;
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is->audio_frame_next_pts = AV_NOPTS_VALUE;
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}
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*pkt_temp = *pkt;
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